We are now running PA with high resolution timers enabled in the kernel. Yet I still can hear the audio glitches. Attached are new traces. On Sun, Jul 24, 2011 at 7:05 PM, Vallabha Hampiholi <vallabha.pa@ googlemail.com> wrote: > Dear Experts, > > I would like to know a few more things: > > Please pardon me, as I started working PulseAudio recently and havent > gone through the source code yet, before asking these questions due to > lack of time. > > 1. How is audio data handled between PulseAudio daemon and PulseAudio > clients? What is the minimum data that should be transfrered before > PulseAudio starts sending the audio data to ALSA hardware? > > 2. Is the above determined by the values of fragment-size and number > of fragments in the daemon.conf file? > > 3. I see that the default values of the above parameters are 25 msec > and 4. So does this mean that, PulseAudio does not start processing > data until 100msec of data is available? > > 4. Also from (3) what should be the ideal value for the latency-time > for pulsesink? 100msec or 25msec? > > Thank you in advance. > > -Rgds > Vallabha > > On 7/23/11, Vallabha Hampiholi <vallabha.pa at googlemail.com> wrote: > > Thank you Pierre and Colin. > > > > @Pierre: Should both parameters be set to -1? Upon gst-inspecting the > > pulsesink element, i get following information: > > buffer-time : Size of audio buffer in microseconds > > flags: readable, writable > > Integer64. Range: 1 - 9223372036854775807 > Default: > > 200000 Current: 200000 > > latency-time : Audio latency in microseconds > > flags: readable, writable > > Integer64. Range: 1 - 9223372036854775807 > Default: > > 10000 Current: 10000 > > Seems like these parameters will not accept negative values. > > > > I am working on an embedded environment, and taking the default vaues of > > latency and buffer time results in buffer underrun. > > > > -Rgds > > Vallabha > > On Fri, Jul 22, 2011 at 11:43 PM, pl bossart > > <bossart.nospam at gmail.com>wrote: > > > >> > Can anyone let me know as what criterions should be considered while > >> setting > >> > the latency-time and buffer-time parameters for the GST element > >> pulsesink? > >> > >> The parameter names are a bit misleading. > >> latency-time only deals with the amount of data exchanged between > >> pulsesink and pulseaudio. It doesn't really represent the latency. > >> buffer-time should be the total buffering/latency you want for your > >> audio chain. If you don't care about it, set it to -1 to reduce the > >> number of wakes and decrease power consumption > >> -Pierre > >> _______________________________________________ > >> pulseaudio-discuss mailing list > >> pulseaudio-discuss at lists.freedesktop.org > >> http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss > >> > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.freedesktop.org/archives/pulseaudio-discuss/attachments/20110727/59d1993b/attachment-0001.htm> -------------- next part -------------- N: main.c: Running in system mode, forcibly disabling SHM mode! N: main.c: Running in system mode, forcibly disabling exit idle time! D: core-rtclock.c: Timer slack is set to 50 us. D: core-util.c: setpriority() worked. I: core-util.c: Successfully gained nice level -11. I: main.c: Found user 'pulse' (UID 109) and group 'pulse' (GID 114). W: main.c: Home directory of user 'pulse' is not '/var/run/pulse', ignoring. I: main.c: Successfully dropped root privileges. I: main.c: This is PulseAudio 0.9.22 D: main.c: Compilation host: arm-none-linux-gnueabi D: main.c: Compilation CFLAGS: -fexpensive-optimizations -fomit-frame-pointer -frename-registers -O2 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option D: main.c: Running on host: Linux armv7l 2.6.37-rc3 #1 Wed Jul 27 09:55:12 IST 2011 D: main.c: Found 1 CPUs. I: main.c: Page size is 4096 bytes D: main.c: Compiled with Valgrind support: no D: main.c: Running in valgrind mode: no D: main.c: Running in VM: no D: main.c: Optimized build: yes D: main.c: All asserts enabled. I: main.c: Machine ID is 89908ad0f00641e5ec7d87d200000002. I: main.c: Using runtime directory /var/run/pulse. I: main.c: Using state directory /var/lib/pulse. I: main.c: Using modules directory /usr/lib/pulse-0.9.22/modules. I: main.c: Running in system mode: yes W: main.c: OK, so you are running PA in system mode. Please note that you most likely shouldn't be doing that. W: main.c: If you do it nonetheless then it's your own fault if things don't work as expected. W: main.c: Please read http://pulseaudio.org/wiki/WhatIsWrongWithSystemMode for an explanation why system mode is usually a bad idea. W: pid.c: Stale PID file, overwriting. I: main.c: Fresh high-resolution timers available! Bon appetit! I: cpu-arm.c: CPU flags: V6 V7 VFP EDSP NEON VFPV3 I: svolume_arm.c: Initialising ARM optimized functions. D: memblock.c: Using private memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65496 D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... D: alsa-util.c: Managed to open hw:0 D: alsa-util.c: Maximum hw buffer size is 743 ms D: alsa-util.c: Set buffer size first (to 88200 samples), period size second (to 88200 samples). I: alsa-sink.c: Successfully opened device hw:0. I: alsa-sink.c: Successfully enabled mmap() mode. I: alsa-sink.c: Successfully enabled timer-based scheduling mode. I: sink.c: Created sink 0 "alsa_output.hw_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: sink.c: alsa.resolution_bits = "16" I: sink.c: device.api = "alsa" I: sink.c: device.class = "sound" I: sink.c: alsa.class = "generic" I: sink.c: alsa.subclass = "generic-mix" I: sink.c: alsa.name = "" I: sink.c: alsa.id = "AIC3X tlv320aic3x-hifi-0" I: sink.c: alsa.subdevice = "0" I: sink.c: alsa.subdevice_name = "subdevice #0" I: sink.c: alsa.device = "0" I: sink.c: alsa.card = "0" I: sink.c: alsa.card_name = "TI81XX EVM" I: sink.c: alsa.long_card_name = "TI81XX EVM" I: sink.c: device.bus_path = "platform-soc-audio.0" I: sink.c: sysfs.path = "/devices/platform/soc-audio.0/sound/card0" I: sink.c: device.string = "hw:0" I: sink.c: device.buffering.buffer_size = "131072" I: sink.c: device.buffering.fragment_size = "8192" I: sink.c: device.access_mode = "mmap+timer" I: sink.c: device.description = "TI81XX EVM" I: sink.c: device.icon_name = "audio-card" I: source.c: Created source 0 "alsa_output.hw_0.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: source.c: device.description = "Monitor of TI81XX EVM" I: source.c: device.class = "monitor" I: source.c: device.icon_name = "audio-input-microphone" I: alsa-sink.c: Using 16.0 fragments of size 8192 bytes (46.44ms), buffer size is 131072 bytes (743.04ms) I: alsa-sink.c: Time scheduling watermark is 20.00ms D: alsa-sink.c: hwbuf_unused=0 D: alsa-sink.c: setting avail_min=31886 D: alsa-util.c: snd_pcm_dump(): D: alsa-util.c: Hardware PCM card 0 'TI81XX EVM' device 0 subdevice 0 D: alsa-util.c: Its setup is: D: alsa-util.c: stream : PLAYBACK D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 32768 D: alsa-util.c: period_size : 2048 D: alsa-util.c: period_time : 46439 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 31886 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 1073741824 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 1073741824 D: alsa-util.c: appl_ptr : 0 D: alsa-util.c: hw_ptr : 0 D: alsa-sink.c: Thread starting up D: core-util.c: SCHED_RR|SCHED_RESET_ON_FORK worked. I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 5. I: alsa-sink.c: Starting playback. D: alsa-sink.c: Cutting sleep time for the initial iterations by half. D: alsa-sink.c: Cutting sleep time for the initial iterations by half. D: alsa-sink.c: Cutting sleep time for the initial iterations by half. I: module.c: Loaded "module-alsa-sink" (index: #0; argument: "device=hw:0"). I: module.c: Loaded "module-native-protocol-unix" (index: #1; argument: "auth-anonymous=1"). I: module.c: Loaded "module-always-sink" (index: #2; argument: ""). D: dbus-util.c: Successfully connected to D-Bus system bus ec9628d92c466a4721584eb100000004 as :1.3 W: main.c: Failed to acquire org.pulseaudio.Server: org.freedesktop.DBus.Error.AccessDenied: Connection ":1.3" is not allowed to own the service "org.pulseaudio.Server" due to security policies in the configuration file I: main.c: Daemon startup complete. I: client.c: Created 0 "Native client (UNIX socket client)" I: protocol-native.c: Client authenticated anonymously. D: protocol-native.c: Protocol version: remote 16, local 16 D: protocol-native.c: SHM possible: no D: protocol-native.c: Negotiated SHM: no D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: sink-input.c: Created input 0 "Playback Stream" on alsa_output.hw_0 with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: sink-input.c: media.name = "Playback Stream" I: sink-input.c: application.name = "gst-launch-0.10" I: sink-input.c: native-protocol.peer = "UNIX socket client" I: sink-input.c: native-protocol.version = "16" I: sink-input.c: application.process.id = "677" I: sink-input.c: application.process.user = "root" I: sink-input.c: application.process.host = "omap3evm" I: sink-input.c: application.process.binary = "gst-launch-0.10" I: sink-input.c: application.language = "C" I: sink-input.c: application.process.machine_id = "89908ad0f00641e5ec7d87d200000002" I: protocol-native.c: Requested tlength=2000.00 ms, minreq=250.00 ms D: protocol-native.c: Traditional mode enabled, modifying sink usec only for compat with minreq. D: memblockq.c: memblockq requested: maxlength=705600, tlength=352800, base=4, prebuf=308700, minreq=44100 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=705600, tlength=352800, base=4, prebuf=308700, minreq=44100 maxrewind=0 I: protocol-native.c: Final latency 2743.04 ms = 1500.00 ms + 2*250.00 ms + 743.04 ms D: alsa-sink.c: Latency set to 743.04ms D: alsa-sink.c: hwbuf_unused=4 D: alsa-sink.c: setting avail_min=31887 D: alsa-sink.c: Requesting rewind due to latency change. D: alsa-sink.c: Requested to rewind 131072 bytes. D: alsa-sink.c: Limited to 130520 bytes. D: alsa-sink.c: before: 32630 D: alsa-sink.c: after: 32630 D: alsa-sink.c: Rewound 130520 bytes. D: sink.c: Processing rewind... D: sink-input.c: Have to rewind 130520 bytes on render memblockq. D: source.c: Processing rewind... D: protocol-native.c: Requesting rewind due to end of underrun. D: alsa-sink.c: Requested to rewind 131072 bytes. D: alsa-sink.c: Limited to 108388 bytes. D: alsa-sink.c: before: 27097 D: alsa-sink.c: after: 27097 D: alsa-sink.c: Rewound 108388 bytes. D: sink.c: Processing rewind... D: sink-input.c: Have to rewind 108388 bytes on render memblockq. D: source.c: Processing rewind...
